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Messages posted by: JohnBiggs
Forum Index » Profile for JohnBiggs » Messages posted by JohnBiggs
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it is something like what you describe. It will not present a full page - I believe - always allowing some space for you to add new entries. There may be a small group of forms that expect you to enter new data on page 2 though, I can not be exhaustive in my answer.. too many forms, not enough time.
Such a guide would be massive (IOW, no). Use the SAT's behavior as your guide. I believe it will show the last few entries as empty where you would put the next item you were adding if you were using the SAT. Follow its lead with your code. So something like entry 5 or 10 is probably blank on the SAT form and I believe you should use that blank entry to add a new entry with. CM will re-sort the list the next time it shows it to you and only provide N entries in its output each time. Keep in mind you code is simulating/acting as the SAT form, not the actual table in CM's database. I know that is less than satisfying answer, but we work with what we are given as far as documentation goes.
Have you
1) created an entry using the CM SAT (successfully)
2) used SMS to 'read' that entry
3) taken the SMS response and altered it slightly to create a new entry

?

I wouldn't expect SMS to have much intelligence here unless the PHP is simply not handling the case you are sending it (net="ext").
the hotfix ONLY applies to the release identified in the PSN (a single release).
It is always best to start a new thread when switching topics. That way all watchers of the forum are notified of the new thread. When you piggy-back on an existing thread only the people watching that thread see your post.

Assuming you are using DMCC, TSAPI or JTAPI, the ConnectionCleared event gives you a reference to the party that is initiating the disconnect activity. To receive that event the application needs to have an active station monitor (in DMCC active call control device monitor) on a extension involved in the call.
The statement regarding SIP Softphones not supporting CTI call control is out of date. You should be able to do call control for Avaya Workplace and one-X. The underlying issue is overall clarity about CTI support by a particular softphone is not in a centralized place within Avaya and it has changed over the years (releases) of the products.

a cos/cor value of 1 merely says you are using configuration number 1 it doesn't tell us how that configuration is setup (which is totally customizable). The error message leads one to think there is some restriction in place there are a number of places to look.
The security database on AE Services may be enabled
The class of restriction of the calling and called parties may block a call from being made between them

An agent login request doesn't really have a called party involved, but sometimes one chooses a similar error message from the list as opposed to creating a new one.

However if CTI call control services are not working in general, one needs to understand why that is. Martin listed the normal places to check, has that been done?
is what you are looking for available here: https://www.devconnectprogram.com/site/global/products_resources/oceana/releases/3_8_1/index.gsp#tabs-product-downld-1
if you look closely (at the monitor ID), you will find that one notification is to the monitor on station 1, and the other is for the monitor on station 2. Events are related to calls, and both stations (and both monitors) 'see' the call from station 1 to station 2.
Have you tried your application against our remote lab (which is configured and works for SIP extensions/stations)?
https://www.devconnectprogram.com/site/global/development_tools_configurations/remote_labs/index.gsp#tabs-product-horiz-3

Are you successfully putting a call control monitor (listener) on the extension?
Are you getting error codes back to your call control requests through DMCC? Have you investigated their meaning using the TSAPI for Communication Manager Programmer's guide?

My root guess here is in your lab the stations are not appropriately configured for CTI on SIP stations.
However, you said you can not use the feature access code to log the SIP agent in - correct? That would eliminate a CTI issue - at least for that problem.

Is the COR/COS for the station extension you are working with allowed to access agent functionality? There is a flag somewhere that controls that if I recall properly.
I am unaware of any means to disable Conference / Transfer / Hold or Drop functions. There are states that the active call can be in where they are not functional (will receive denial behavior if used). Please describe how one would configure a station so that call control functionality would be denied at all times to all destinations. The applications that I am aware of that do not present certain call control buttons at various moments in a call flow are using the state information that Martian mentioned to drive that behavior.
you used the phrase "make call" and perhaps I fell into a trap. I presumed underneath your method call triggered a particular station to originate a call to your destination number. It dawned on me later that I don't really know what the services on Breeze are using as an originating device. let me ask another pair of eyes to look at your problem statement. It would be helpful to know exactly what method you are using to originate calls with.
I am not exactly sure what services Breeze makes available to you. I am familiar with Communication Services (basically this is exposing TSAPI).

A station device (Communication Manager Station Extension) can only originate one call at a time, unless you place the call on hold, and even then you are limited by the number of call appearances provisioned on that station (max 10).

I suspect you are running into that limitation, but it is not obvious given your 'what am I doing' description.

If you are using a SIP trunk in your application, you don't have to go through the complexity of using stations on communication manager as your originating device, but you will still be limited by the number of members in the trunk group.

I believe you are going to have to register a DMCC station to be used as the originator for each call (or block of 10 calls) you are placing, which means you need a list of station extensions to launch calls from, or use System Management Service to create stations (and remove them when your task completes).
Captcha - try a different browser/ update your browser version? Not sure what that problem may be, but you could open a General Support ticket to get help on that. https://www.devconnectprogram.com/ticket/generalTicket

what kind of quality of service stats? RTCP? or speed of service by agents and queue wait times? something else? Is that a SAT screen that provides the information you are hoping to get?
The answer varies by release of AE Services. It can be found in the 'capacities' section of the AES Overview document for that release

https://www.devconnectprogram.com/site/global/products_resources/avaya_aura_application_enablement_services/releases/8_1_3/index.gsp#tabs-product-downld-2

For 8.1 the maxima is 8,000 active DMCC device registrations per session. However, there are considerations use of encryption, use of server media, and how frequently can a registration be done for one), is dependent on other factors which need to be taken into account in any application that scales. So what you will achieve in your environment requires more careful examination of the system configuration than can be done through a forum exchange.
Is your question
1) how many sessions can be active, or
2) how many active registrations per session, or
3) how many inactive registrations (i.e. DeviceIDs) can a single session have?
 
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